U.S. Philips Corporation

Bernd Selbach

Peter Vary

Peter Hessler

Manfred Schmidt

Michael Behrens

A circuit arrangement for establishing conference connections comprising conference units is described, which units are linked to form a loop (L). In this loop (L) sum codewords consisting of sample values of the signals of all participants in a conference are transmitted from one conference unit to the next. In each conference unit a sum codeword is updated so that the sample value of a conference signal from the previous loop travel is replaced by the current sample value. In order to enable the monotoring of a travel of sum codewords through the loop with a circuit arrangement of this type and in order to have the sum codewords pass through the loop in a predetermined brief period of time without a necessity for a synchronization of individual processing modules of the conference units, the function of one or more conference units is taken over by a programmed processor (P1 to P4). The exchange of data between the conferees and the processors is controlled by means of an interface circuit (PO). In addition to the sum codewords also test words consisting of transformed sum codewords pass through the loop (L). Sum codewords and test words are buffered in input memories (ES1 to ES4) during their travel through the loop until the associated processor (P1 to P4) has finished the program modules for which the sum codeword or test word respectively is used. These modules are programmed in the individual processors (P1 to P4) in a time offset order.

Circuit arrangement for establishing conference connections

A circuit arrangement for channel-specific processing of a multi-channel input signal which may consist of two types of encoded signals, a first type wherein all channel signals are encoded by a bit-oriented method (e.g., delta modulation) and a second type wherein all channel signals are encoded by a character-oriented method (e.g., PCM). An interface process (SSP), to which the input signal is applied, is coupled to a bidirectional interface (BIS) for signals of the first type and which is controllable by means of write and read addresses, and is also coupled to at least two such bidirectionally controllable interfaces (BI0, BI1, . . . BI7) for signals of the second type. The latter interfaces are respectively coupled to respective transcoder processors (AP0, AP1, . . . AP7). One or more data outputs of the interface (BIS) for signals of the first type are respectively coupled to respective ones of the transcoder processors (AP0, AP1, . . . AP7). The transcoded channel signals produced by such processors are transmitted via the interfaces (BIS, BI0, . . . BI7) to the interface processor (SSP), which combines them into a multi-channel output signal.

Circuit arrangement for channel-specific processing of a multi-channel

Method of recognizing pauses in a speech signal when a slowly varying noise signal is superposed on the speech signal. For the purpose of pause recognition so-called short-time mean values connected with a clock pulse are continuously determined from the samples of the disturbed speech signal, which short time mean values are a measure of the average power of approximately 100 ms long sections of the disturbed speech signals. The sequence of these short-time mean values is then smoothed by linear filtration or by means of a median filter. In parallel with the smoothing operation an estimate for the noise signal power averaged over a few seconds is taken from the sequence of short-time mean values. If the smoothed short-time mean value is once or several times less than a threshold which is proportional to the above-mentioned estimate, then it is decided that there is a speech pause.

Method of recognizing speech pauses

A method of recognizing speech pauses in a speech signal even when the signal is disturbed by a slowly varying noise signal superposed thereon. Mean values which are an approximate measure of the average power of successive sections of the disturbed signal are determined from the short-time Fourier coefficients of the disturbed speech signal. The sequential short-time mean values are then smoothed by a linear digital filter or a median filter. An estimate of the noise signal power averaged over a few seconds is also recovered from the sequence of short-time mean values. A speech pause is signified when the smoothed short-time mean value (output of GL) more than once falls to a threshold which is proportional to the estimated noise power (output of PA).

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Eckental, Germany

Bernd Selbach